| Category | Q&A |
 Setup Q&A
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The space of Line Setting / Line Number / should be left blank, and choose "Register Number" on Line Feature / When VoIP call out / Send ANI by /, then Caller ID will be display on Display name of Port Status.
Please cancel the check box of "rompt Voice"of VoIP Setup / Advance Setup / Promot Voice / for VoIP call in.
rs: represents Register Server, can make phone call out automatically by register to Server successfully, the number order is 1_2_3_4.
rs1: represents to route the call via Register Server of number 1, rs2 represents to route the call via Register Sever of number 2, and so on.
rs1_3: represents the priority of routing the call is via rs1; if the Server registered of number 1 is failed that will route the call via rs3, and so on.
Pstn: represents to route the call via PSTN line interface of the device applied to G3000C series, G111b and other VoIP devices's spec. including PSTN port.
sip:ip:UDP: represents the call communication is via peer to peer of SIP protocol, UDP represents the Listen port of receiver.
h323:ip:UDP: represents the call communication via peer to peer of H.323 protocol, UDP represents the Listen port of receiver.
Ipivr: Enter the Network parameter voice interactive setting mode. Users can us this function to enter all the WAN network parameters without PC. (Please refer to the application note "IP IVR procedure" on User's Manual for more detail).
Rect: Enter to voice record procedure. Users can assign a function code for enter the voice record procedure, when press this code to enter the voice record procedure, the device will record 30 seconds voice file and keep on sound wave file (G.711, uLaw), Users can download the recorded wave file on / VoIP Setup / Advance setting / Prompt Voice / and use this file to upload for customization voice or use for busy tone analysis.
lo: assign the route to local loop back. The destination IP of this call will be the local host, i.e.: 127.0.0.1.
The space also can be configured IP address or Domain Name for communication of peer to peer, also assigned the communication port later.
rs1: represents to route the call via Register Server of number 1, rs2 represents to route the call via Register Sever of number 2, and so on.
rs1_3: represents the priority of routing the call is via rs1; if the Server registered of number 1 is failed that will route the call via rs3, and so on.
Pstn: represents to route the call via PSTN line interface of the device applied to G3000C series, G111b and other VoIP devices's spec. including PSTN port.
sip:ip:UDP: represents the call communication is via peer to peer of SIP protocol, UDP represents the Listen port of receiver.
h323:ip:UDP: represents the call communication via peer to peer of H.323 protocol, UDP represents the Listen port of receiver.
Ipivr: Enter the Network parameter voice interactive setting mode. Users can us this function to enter all the WAN network parameters without PC. (Please refer to the application note "IP IVR procedure" on User's Manual for more detail).
Rect: Enter to voice record procedure. Users can assign a function code for enter the voice record procedure, when press this code to enter the voice record procedure, the device will record 30 seconds voice file and keep on sound wave file (G.711, uLaw), Users can download the recorded wave file on / VoIP Setup / Advance setting / Prompt Voice / and use this file to upload for customization voice or use for busy tone analysis.
lo: assign the route to local loop back. The destination IP of this call will be the local host, i.e.: 127.0.0.1.
The space also can be configured IP address or Domain Name for communication of peer to peer, also assigned the communication port later.
hl: represents Hot-Line, the port number would be added behind hl, such as if the call made through port 3 that would be filled in as hl3, and so on.
hl1: configured port 1 for Hot-Line, and so on.
(Please choose "Enable" on Line Setting / Hotline / for above two setting).
v: if the call route to default Server is disable, then the call will route to PSTN line by this rule.
v0: if the call route to default Server with the beginning number of 0 is disable, then the call will route to PSTN line by this rule, and so on.
p: if the call out via the assigned PSTN line is disable, then the call out will route to the assigned Sever by this rule, and so on.
p1: if the call route via PSTN with the beginning number of 1, then the call out will route to the assigned Server, and so on.
(These above four prefix code should be combined with the setting in pstn / rs of the Destination column used for G3000C series and G111b only right now).
hl1: configured port 1 for Hot-Line, and so on.
(Please choose "Enable" on Line Setting / Hotline / for above two setting).
v: if the call route to default Server is disable, then the call will route to PSTN line by this rule.
v0: if the call route to default Server with the beginning number of 0 is disable, then the call will route to PSTN line by this rule, and so on.
p: if the call out via the assigned PSTN line is disable, then the call out will route to the assigned Sever by this rule, and so on.
p1: if the call route via PSTN with the beginning number of 1, then the call out will route to the assigned Server, and so on.
(These above four prefix code should be combined with the setting in pstn / rs of the Destination column used for G3000C series and G111b only right now).
Virtual Server can be assigned to one of LANs' IP and port mapping to one of port of WANs' IP(such as below diagram that LAN IP 192.168.22.123 of port 5689 mapping to WAN IP's port 5689).
Port Mapping can be assigned to one of LANs' IP mapping to one of port or multiple ports of WANs' IP. As below diagram, Mapping Ports Mapping Ports can be used " - " or " , " to separate the multiple ports.
Port Mapping can be assigned to one of LANs' IP mapping to one of port or multiple ports of WANs' IP. As below diagram, Mapping Ports Mapping Ports can be used " - " or " , " to separate the multiple ports.
Please click the "Browse" button of System Setup / Backup / Restore / Configurations / of Web UI to select the back configuration parameters file, then click "Restore" button to upload it; after that, please click "Reboot" of System Setup / Backup / Restore / Configurations / Reboot System / for completing the Restore function.
Please note: In order not to save the old date, please don't execute the action of "Save".
Please note: In order not to save the old date, please don't execute the action of "Save".
The prefix number will be added on the users' press number, e.g. if users press the number is "567" and the prefix field is set to "123", then the actual call out number will be "123567"; if the prefix field is set to "@123", then the actual call out number will be "567123".
Define each item on Forwarding below.
Name: Specify a profile name. Please use UNDERLINE to instead of SPACE due to the HTTP protocol limitation.
Always: Always redirect forward to this IP(or URL) / Phone number, all incoming call will be forwarded to the assigned IP here.
On Busy: Redirect forward to this IP(or URL)/Phone number when busy, all incoming VoIP call will be forwarded to the assigned IP here when this line is busy.
No Answer:Redirect forward to this IP(or URL)/Phone number when no answer over the time No Answer Sec, an incoming VoIP call will be forwarded to the assigned IP here when ring time over the defined No Answer Sec.
Below is the Configuration method for users want to forward the call to the IP of 172.16.39.2 for calling to 66390001 directly, please also revise the call number to 63986399.
Configuration on VoIP Call in below:
Forwarding:
Name: Specify a profile name. Please use UNDERLINE to instead of SPACE due to the HTTP protocol limitation.
Always: Always redirect forward to this IP(or URL) / Phone number, all incoming call will be forwarded to the assigned IP here.
On Busy: Redirect forward to this IP(or URL)/Phone number when busy, all incoming VoIP call will be forwarded to the assigned IP here when this line is busy.
No Answer:Redirect forward to this IP(or URL)/Phone number when no answer over the time No Answer Sec, an incoming VoIP call will be forwarded to the assigned IP here when ring time over the defined No Answer Sec.
Below is the Configuration method for users want to forward the call to the IP of 172.16.39.2 for calling to 66390001 directly, please also revise the call number to 63986399.
Configuration on VoIP Call in below:
Forwarding:
When this gateway has been used for outbound call, it can be enabled to check the remote caller gateway's IP to decide to accept or refuse the call. If define the IP range here and enable the "Auth" option on the / VoIP Setup / Routing Setup / VoIP Call in /, only the IP in range will be allowed to call out by this gateway.
Port Status display the current and last time of VoIP call status & result, and can be renewed the status by the "Reload" function, but all the record will be deleted if reboot the device.
Port : Display which port is on connection status, e.g. 1 or 2.
Display name: Display the remote party name of this VoIP call.
Status: Current status of this port.
Idle: Standby for making a phone call.
Signal: Waiting for DTMF press or VoIP protocol connecting.
In: There is a phone call made from phone port and call out to Network by VoIP.
Out: There is a phone call made from Network VoIP and pick up by phone set.
Connected IP: The remote party's IP of this VoIP call.
Caller ID: Caller ID received from telephone line port.
Start time: Date & Time of this VoIP begins to make a call on this port, the time is calculated from the phone rings.
End time: Date & Time of making last VoIP call on this port.
Talking Sec: Total talking seconds of last VoIP call on this port.
Dialed number:
One the VoIP call out (line status display"In"). This will display the real dial out number for VoIP call.
On the VoIP call in (line status display"out"). This will display the number will dial out to phone line.
Release by: This will display the terminated reason of this call.
(134)onHangup: Represents the local party is for calling out, and as P2P or the call can be transferred via system only; the call cannot be gone through for speaking and hung up by local or remote party.
(212)H323Release: Represents the local party is for calling in (receiver), the call cannot be gone through for speaking and hung up by local or remote party.
(107)inDigitLess: Represents the call number is not enough.
(142)VoipRelease: Represents the local party is for calling out, the call can be gone through for speaking and hung up by the receiver.
(236)onHangup: Represents the local party is for calling in (receiver), the call can be gone through for speaking and hung up by the local party.
(106)noAreaCode: Represents no any area code to correspond with the call.
(132)H323Release: Represents the local party is for calling out, and the call can be transferred by the system; but the system responses the call out number is wrong and hang it up.
(105)inDigitOnHangup: Represents the call is hung up before completing the dial number.
(123)AdmissionFailure: Represents the local party is for calling out, and the call can be transferred by the system; but the system responses not to find the corresponding call out number and hang the call up.
(103)inDigitOnHangup: Represents the call is no any response on remote party.
(146)onHangup: Represents the local party is for calling out, and as P2P or the call can be transferred via system only; the call cannot be gone through for speaking and hung up by local party.
(232)VoipRelease: Represents the local party is for calling in (receiver), the call can be gone through for speaking and hung up by remote party.
(222)VoipRelease: Represents the local party is for calling in (receiver), the call can not be gone through for speaking and hung up by remote party.
(102)inDigitOnMaxTime: Represents the local party is for calling out and not to complete the dialing number before the standard Max Time.
(121)VoipFaliure: Represents the route of VoIP Call out can not be called out. (If the configured route is ineffective or wrong).
Port : Display which port is on connection status, e.g. 1 or 2.
Display name: Display the remote party name of this VoIP call.
Status: Current status of this port.
Idle: Standby for making a phone call.
Signal: Waiting for DTMF press or VoIP protocol connecting.
In: There is a phone call made from phone port and call out to Network by VoIP.
Out: There is a phone call made from Network VoIP and pick up by phone set.
Connected IP: The remote party's IP of this VoIP call.
Caller ID: Caller ID received from telephone line port.
Start time: Date & Time of this VoIP begins to make a call on this port, the time is calculated from the phone rings.
End time: Date & Time of making last VoIP call on this port.
Talking Sec: Total talking seconds of last VoIP call on this port.
Dialed number:
One the VoIP call out (line status display"In"). This will display the real dial out number for VoIP call.
On the VoIP call in (line status display"out"). This will display the number will dial out to phone line.
Release by: This will display the terminated reason of this call.
(134)onHangup: Represents the local party is for calling out, and as P2P or the call can be transferred via system only; the call cannot be gone through for speaking and hung up by local or remote party.
(212)H323Release: Represents the local party is for calling in (receiver), the call cannot be gone through for speaking and hung up by local or remote party.
(107)inDigitLess: Represents the call number is not enough.
(142)VoipRelease: Represents the local party is for calling out, the call can be gone through for speaking and hung up by the receiver.
(236)onHangup: Represents the local party is for calling in (receiver), the call can be gone through for speaking and hung up by the local party.
(106)noAreaCode: Represents no any area code to correspond with the call.
(132)H323Release: Represents the local party is for calling out, and the call can be transferred by the system; but the system responses the call out number is wrong and hang it up.
(105)inDigitOnHangup: Represents the call is hung up before completing the dial number.
(123)AdmissionFailure: Represents the local party is for calling out, and the call can be transferred by the system; but the system responses not to find the corresponding call out number and hang the call up.
(103)inDigitOnHangup: Represents the call is no any response on remote party.
(146)onHangup: Represents the local party is for calling out, and as P2P or the call can be transferred via system only; the call cannot be gone through for speaking and hung up by local party.
(232)VoipRelease: Represents the local party is for calling in (receiver), the call can be gone through for speaking and hung up by remote party.
(222)VoipRelease: Represents the local party is for calling in (receiver), the call can not be gone through for speaking and hung up by remote party.
(102)inDigitOnMaxTime: Represents the local party is for calling out and not to complete the dialing number before the standard Max Time.
(121)VoipFaliure: Represents the route of VoIP Call out can not be called out. (If the configured route is ineffective or wrong).
